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-rw-r--r--NEWS7
-rw-r--r--src/decoder/wavpack_decoder_plugin.c47
-rw-r--r--src/encoder/vorbis_encoder.c11
-rw-r--r--src/encoder_plugin.h24
-rw-r--r--src/output/alsa_plugin.c8
-rw-r--r--src/output/httpd_output_plugin.c5
-rw-r--r--src/output/shout_plugin.c2
-rw-r--r--src/output_thread.c7
-rw-r--r--src/pipe.c5
-rw-r--r--src/pipe.h5
-rw-r--r--src/player_thread.c2
-rw-r--r--src/playlist_control.c8
-rw-r--r--src/update_walk.c10
13 files changed, 102 insertions, 39 deletions
diff --git a/NEWS b/NEWS
index adbad0484..0698f5d1c 100644
--- a/NEWS
+++ b/NEWS
@@ -21,9 +21,16 @@ ver 0.17 (2011/??/??)
ver 0.16.4 (2011/??/??)
* fix memory leaks
+* don't resume playback when seeking to another song while paused
+* apply follow_inside_symlinks to absolute symlinks
* decoder:
- ffmpeg: workaround for semantic API change in recent ffmpeg versions
- flac: validate the sample rate when scanning the tag
+ - wavpack: obey all decoder commands, stop at CUE track border
+* encoder:
+ - vorbis: don't send end-of-stream on flush
+* output:
+ - alsa: fix SIGFPE when alsa announces a period size of 0
ver 0.16.3 (2011/06/04)
diff --git a/src/decoder/wavpack_decoder_plugin.c b/src/decoder/wavpack_decoder_plugin.c
index b14a41f93..200bf6455 100644
--- a/src/decoder/wavpack_decoder_plugin.c
+++ b/src/decoder/wavpack_decoder_plugin.c
@@ -34,9 +34,6 @@
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "wavpack"
-/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
-#define CHUNK_SIZE 1020
-
#define ERRORLEN 80
static struct {
@@ -162,8 +159,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
enum sample_format sample_format;
struct audio_format audio_format;
format_samples_t format_samples;
- char chunk[CHUNK_SIZE];
- int samples_requested, samples_got;
float total_time;
int bytes_per_sample, output_sample_size;
@@ -193,12 +188,15 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
output_sample_size = audio_format_frame_size(&audio_format);
/* wavpack gives us all kind of samples in a 32-bit space */
- samples_requested = sizeof(chunk) / (4 * audio_format.channels);
+ int32_t chunk[1024];
+ const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
+ audio_format.channels;
decoder_initialized(decoder, &audio_format, can_seek, total_time);
- do {
- if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+ enum decoder_command cmd = decoder_get_command(decoder);
+ while (cmd != DECODE_COMMAND_STOP) {
+ if (cmd == DECODE_COMMAND_SEEK) {
if (can_seek) {
unsigned where = decoder_seek_where(decoder) *
audio_format.sample_rate;
@@ -213,29 +211,20 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
}
}
- if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
+ uint32_t samples_got = WavpackUnpackSamples(wpc, chunk,
+ samples_requested);
+ if (samples_got == 0)
break;
- }
- samples_got = WavpackUnpackSamples(
- wpc, (int32_t *)chunk, samples_requested
- );
- if (samples_got > 0) {
- int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
- 1000 + 0.5);
-
- format_samples(
- bytes_per_sample, chunk,
- samples_got * audio_format.channels
- );
-
- decoder_data(
- decoder, NULL, chunk,
- samples_got * output_sample_size,
- bitrate
- );
- }
- } while (samples_got > 0);
+ int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
+ 0.5);
+ format_samples(bytes_per_sample, chunk,
+ samples_got * audio_format.channels);
+
+ cmd = decoder_data(decoder, NULL, chunk,
+ samples_got * output_sample_size,
+ bitrate);
+ }
}
/**
diff --git a/src/encoder/vorbis_encoder.c b/src/encoder/vorbis_encoder.c
index 5b3110988..3e9d486b6 100644
--- a/src/encoder/vorbis_encoder.c
+++ b/src/encoder/vorbis_encoder.c
@@ -266,6 +266,15 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
{
struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+ encoder->flush = true;
+ return true;
+}
+
+static bool
+vorbis_encoder_pre_tag(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
+{
+ struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
vorbis_analysis_wrote(&encoder->vd, 0);
vorbis_encoder_blockout(encoder);
@@ -366,6 +375,7 @@ vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
if (ret == 0 && encoder->flush) {
encoder->flush = false;
ret = ogg_stream_flush(&encoder->os, &page);
+
}
if (ret == 0)
@@ -398,6 +408,7 @@ const struct encoder_plugin vorbis_encoder_plugin = {
.open = vorbis_encoder_open,
.close = vorbis_encoder_close,
.flush = vorbis_encoder_flush,
+ .pre_tag = vorbis_encoder_pre_tag,
.tag = vorbis_encoder_tag,
.write = vorbis_encoder_write,
.read = vorbis_encoder_read,
diff --git a/src/encoder_plugin.h b/src/encoder_plugin.h
index 4a6e3805e..95b3da016 100644
--- a/src/encoder_plugin.h
+++ b/src/encoder_plugin.h
@@ -50,6 +50,8 @@ struct encoder_plugin {
bool (*flush)(struct encoder *encoder, GError **error);
+ bool (*pre_tag)(struct encoder *encoder, GError **error);
+
bool (*tag)(struct encoder *encoder, const struct tag *tag,
GError **error);
@@ -148,8 +150,30 @@ encoder_flush(struct encoder *encoder, GError **error)
}
/**
+ * Prepare for sending a tag to the encoder. This is used by some
+ * encoders to flush the previous sub-stream, in preparation to begin
+ * a new one.
+ *
+ * @param encoder the encoder
+ * @param tag the tag object
+ * @param error location to store the error occuring, or NULL to ignore errors.
+ * @return true on success
+ */
+static inline bool
+encoder_pre_tag(struct encoder *encoder, GError **error)
+{
+ /* this method is optional */
+ return encoder->plugin->pre_tag != NULL
+ ? encoder->plugin->pre_tag(encoder, error)
+ : true;
+}
+
+/**
* Sends a tag to the encoder.
*
+ * Instructions: call encoder_pre_tag(); then obtain flushed data with
+ * encoder_read(); finally call encoder_tag().
+ *
* @param encoder the encoder
* @param tag the tag object
* @param error location to store the error occurring, or NULL to ignore errors.
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 1c8c7b7f7..0bbe231fd 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -508,6 +508,14 @@ configure_hw:
g_debug("buffer_size=%u period_size=%u",
(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
+ if (alsa_period_size == 0)
+ /* this works around a SIGFPE bug that occurred when
+ an ALSA driver indicated period_size==0; this
+ caused a division by zero in alsa_play(). By using
+ the fallback "1", we make sure that this won't
+ happen again. */
+ alsa_period_size = 1;
+
ad->period_frames = alsa_period_size;
ad->period_position = 0;
diff --git a/src/output/httpd_output_plugin.c b/src/output/httpd_output_plugin.c
index 20098c90e..3f570c7b9 100644
--- a/src/output/httpd_output_plugin.c
+++ b/src/output/httpd_output_plugin.c
@@ -493,7 +493,8 @@ httpd_output_pause(void *data)
if (has_clients) {
static const char silence[1020];
- return httpd_output_play(data, silence, sizeof(silence), NULL);
+ return httpd_output_play(data, silence, sizeof(silence),
+ NULL) > 0;
} else {
g_usleep(100000);
return true;
@@ -522,7 +523,7 @@ httpd_output_tag(void *data, const struct tag *tag)
/* flush the current stream, and end it */
- encoder_flush(httpd->encoder, NULL);
+ encoder_pre_tag(httpd->encoder, NULL);
httpd_output_encoder_to_clients(httpd);
/* send the tag to the encoder - which starts a new
diff --git a/src/output/shout_plugin.c b/src/output/shout_plugin.c
index ec5fca3d9..7a4d70e70 100644
--- a/src/output/shout_plugin.c
+++ b/src/output/shout_plugin.c
@@ -518,7 +518,7 @@ static void my_shout_set_tag(void *data,
if (sd->encoder->plugin->tag != NULL) {
/* encoder plugin supports stream tags */
- ret = encoder_flush(sd->encoder, &error);
+ ret = encoder_pre_tag(sd->encoder, &error);
if (!ret) {
g_warning("%s", error->message);
g_error_free(error);
diff --git a/src/output_thread.c b/src/output_thread.c
index 21096eb04..ec5fc5b31 100644
--- a/src/output_thread.c
+++ b/src/output_thread.c
@@ -641,8 +641,13 @@ static gpointer audio_output_task(gpointer arg)
case AO_COMMAND_CANCEL:
ao->chunk = NULL;
- if (ao->open)
+
+ if (ao->open) {
+ g_mutex_unlock(ao->mutex);
ao_plugin_cancel(ao->plugin, ao->data);
+ g_mutex_lock(ao->mutex);
+ }
+
ao_command_finished(ao);
/* the player thread will now clear our music
diff --git a/src/pipe.c b/src/pipe.c
index db80c2b2d..d8131432f 100644
--- a/src/pipe.c
+++ b/src/pipe.c
@@ -187,5 +187,8 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk)
unsigned
music_pipe_size(const struct music_pipe *mp)
{
- return mp->size;
+ g_mutex_lock(mp->mutex);
+ unsigned size = mp->size;
+ g_mutex_unlock(mp->mutex);
+ return size;
}
diff --git a/src/pipe.h b/src/pipe.h
index 785c1b943..84b9869e0 100644
--- a/src/pipe.h
+++ b/src/pipe.h
@@ -20,6 +20,7 @@
#ifndef MPD_PIPE_H
#define MPD_PIPE_H
+#include <glib.h>
#include <stdbool.h>
#ifndef NDEBUG
@@ -38,6 +39,7 @@ struct music_pipe;
/**
* Creates a new #music_pipe object. It is empty.
*/
+G_GNUC_MALLOC
struct music_pipe *
music_pipe_new(void);
@@ -70,6 +72,7 @@ music_pipe_contains(const struct music_pipe *mp,
* Returns the first #music_chunk from the pipe. Returns NULL if the
* pipe is empty.
*/
+G_GNUC_PURE
const struct music_chunk *
music_pipe_peek(const struct music_pipe *mp);
@@ -96,9 +99,11 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk);
/**
* Returns the number of chunks currently in this pipe.
*/
+G_GNUC_PURE
unsigned
music_pipe_size(const struct music_pipe *mp);
+G_GNUC_PURE
static inline bool
music_pipe_empty(const struct music_pipe *mp)
{
diff --git a/src/player_thread.c b/src/player_thread.c
index d47c5c851..58682f2ca 100644
--- a/src/player_thread.c
+++ b/src/player_thread.c
@@ -628,7 +628,9 @@ play_chunk(struct player_control *pc,
return true;
}
+ player_lock(pc);
pc->bit_rate = chunk->bit_rate;
+ player_unlock(pc);
/* send the chunk to the audio outputs */
diff --git a/src/playlist_control.c b/src/playlist_control.c
index df9384a80..998294845 100644
--- a/src/playlist_control.c
+++ b/src/playlist_control.c
@@ -230,10 +230,12 @@ playlist_seek_song(struct playlist *playlist, struct player_control *pc,
playlist->error_count = 0;
if (!playlist->playing || (unsigned)playlist->current != i) {
- /* seeking is not within the current song - first
- start playing the new song */
+ /* seeking is not within the current song - prepare
+ song change */
+
+ playlist->playing = true;
+ playlist->current = i;
- playlist_play_order(playlist, pc, i);
queued = NULL;
}
diff --git a/src/update_walk.c b/src/update_walk.c
index e5ab4fc37..e70ead173 100644
--- a/src/update_walk.c
+++ b/src/update_walk.c
@@ -714,8 +714,14 @@ skip_symlink(const struct directory *directory, const char *utf8_name)
return false;
}
- if (buffer[0] == '/')
- return !follow_outside_symlinks;
+ if (g_path_is_absolute(buffer)) {
+ /* if the symlink points to an absolute path, see if
+ that path is inside the music directory */
+ const char *relative = map_to_relative_path(buffer);
+ return relative > buffer
+ ? !follow_inside_symlinks
+ : !follow_outside_symlinks;
+ }
p = buffer;
while (*p == '.') {
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 07:23:15 +0000